This patch adds support for the Analog Devices ADAV801 and ADAV803 audio codec.
Signed-off-by: Lars-Peter Clausen <[email protected]>
---
sound/soc/codecs/Kconfig | 4 +
sound/soc/codecs/Makefile | 2 +
sound/soc/codecs/adav80x.c | 846 ++++++++++++++++++++++++++++++++++++++++++++
sound/soc/codecs/adav80x.h | 23 ++
4 files changed, 875 insertions(+), 0 deletions(-)
create mode 100644 sound/soc/codecs/adav80x.c
create mode 100644 sound/soc/codecs/adav80x.h
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 2998e65..ff43405 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -17,6 +17,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI
select SND_SOC_AD1980 if SND_SOC_AC97_BUS
select SND_SOC_AD73311
+ select SND_SOC_ADAV80X
select SND_SOC_ADS117X
select SND_SOC_AK4104 if SPI_MASTER
select SND_SOC_AK4535 if I2C
@@ -137,6 +138,9 @@ config SND_SOC_ADAU1701
select SIGMA
tristate
+config SND_SOC_ADAV80X
+ tristate
+
config SND_SOC_ADS117X
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 51cd3d4..4957431 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -5,6 +5,7 @@ snd-soc-ad193x-objs := ad193x.o
snd-soc-ad1980-objs := ad1980.o
snd-soc-ad73311-objs := ad73311.o
snd-soc-adau1701-objs := adau1701.o
+snd-soc-adav80x-objs := adav80x.o
snd-soc-ads117x-objs := ads117x.o
snd-soc-ak4104-objs := ak4104.o
snd-soc-ak4535-objs := ak4535.o
@@ -99,6 +100,7 @@ obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o
obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o
obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o
+obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o
obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o
obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
new file mode 100644
index 0000000..d2f3d08
--- /dev/null
+++ b/sound/soc/codecs/adav80x.c
@@ -0,0 +1,846 @@
+/*
+ * ADAV80X Audio Codec driver supporting ADAV801, ADAV803
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Yi Li <[email protected]>
+ * Author: Lars-Peter Clausen <[email protected]>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/i2c.h>
+#include <linux/spi/spi.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+
+#include "adav80x.h"
+
+enum adav80x_clk_src {
+ ADAV80X_CLK_XIN ,
+ ADAV80X_CLK_PLL1,
+ ADAV80X_CLK_PLL2,
+ ADAV80X_CLK_INTERNAL1,
+ ADAV80X_CLK_INTERNAL2,
+ ADAV80X_CLK_MCLKI,
+};
+
+#define ADAV80X_PLAYBACK_CTRL 0x04
+#define ADAV80X_AUX_IN_CTRL 0x05
+#define ADAV80X_REC_CTRL 0x06
+#define ADAV80X_AUX_OUT_CTRL 0x07
+#define ADAV80X_DPATH_CTRL1 0x62
+#define ADAV80X_DPATH_CTRL2 0x63
+#define ADAV80X_DAC_CTRL1 0x64
+#define ADAV80X_DAC_CTRL2 0x65
+#define ADAV80X_DAC_CTRL3 0x66
+#define ADAV80X_DAC_L_VOL 0x68
+#define ADAV80X_DAC_R_VOL 0x69
+#define ADAV80X_PGA_L_VOL 0x6c
+#define ADAV80X_PGA_R_VOL 0x6d
+#define ADAV80X_ADC_CTRL1 0x6e
+#define ADAV80X_ADC_CTRL2 0x6f
+#define ADAV80X_ADC_L_VOL 0x70
+#define ADAV80X_ADC_R_VOL 0x71
+#define ADAV80X_PLL_CTRL1 0x74
+#define ADAV80X_PLL_CTRL2 0x75
+#define ADAV80X_ICLK_CTRL1 0x76
+#define ADAV80X_ICLK_CTRL2 0x77
+#define ADAV80X_PLL_CLK_SRC 0x78
+#define ADAV80X_PLL_OUTE 0x7a
+
+#define ADAV80X_PLL_CLK_SRC_PLL_XIN(pll) 0x00
+#define ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll) (0x40 << (pll))
+#define ADAV80X_PLL_CLK_SRC_PLL_MASK(pll) (0x40 << (pll))
+
+#define ADAV80X_PLL_CTRL1_PLLDIV 0x10
+#define ADAV80X_PLL_CTRL1_PLL2PD 0x08
+#define ADAV80X_PLL_CTRL1_PLL1PD 0x04
+#define ADAV80X_PLL_CTRL1_XTLPD 0x02
+
+#define ADAV80X_PLL_CTRL2_FIELD(pll, x) ((x) << ((pll) * 4))
+
+#define ADAV80X_PLL_CTRL2_FS_48(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x00)
+#define ADAV80X_PLL_CTRL2_FS_32(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x08)
+#define ADAV80X_PLL_CTRL2_FS_44(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x0c)
+
+#define ADAV80X_PLL_CTRL2_SEL(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x02)
+#define ADAV80X_PLL_CTRL2_DOUB(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x01)
+#define ADAV80X_PLL_CTRL2_PLL_MASK(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x0f)
+
+#define ADAV80X_ADC_CTRL1_MODULATOR_MASK 0x80
+#define ADAV80X_ADC_CTRL1_MODULATOR_128FS 0x00
+#define ADAV80X_ADC_CTRL1_MODULATOR_64FS 0x80
+
+#define ADAV80X_DAC_CTRL1_PD 0x80
+
+#define ADAV80X_DAC_CTRL2_DIV1 0x00
+#define ADAV80X_DAC_CTRL2_DIV1_5 0x10
+#define ADAV80X_DAC_CTRL2_DIV2 0x20
+#define ADAV80X_DAC_CTRL2_DIV3 0x30
+#define ADAV80X_DAC_CTRL2_DIV_MASK 0x30
+
+#define ADAV80X_DAC_CTRL2_INTERPOL_256FS 0x00
+#define ADAV80X_DAC_CTRL2_INTERPOL_128FS 0x40
+#define ADAV80X_DAC_CTRL2_INTERPOL_64FS 0x80
+#define ADAV80X_DAC_CTRL2_INTERPOL_MASK 0xc0
+
+#define ADAV80X_DAC_CTRL2_DEEMPH_NONE 0x00
+#define ADAV80X_DAC_CTRL2_DEEMPH_44 0x01
+#define ADAV80X_DAC_CTRL2_DEEMPH_32 0x02
+#define ADAV80X_DAC_CTRL2_DEEMPH_48 0x03
+#define ADAV80X_DAC_CTRL2_DEEMPH_MASK 0x01
+
+#define ADAV80X_CAPTURE_MODE_MASTER 0x20
+#define ADAV80X_CAPTURE_WORD_LEN24 0x00
+#define ADAV80X_CAPTURE_WORD_LEN20 0x04
+#define ADAV80X_CAPTRUE_WORD_LEN18 0x08
+#define ADAV80X_CAPTURE_WORD_LEN16 0x0c
+#define ADAV80X_CAPTURE_WORD_LEN_MASK 0x0c
+
+#define ADAV80X_CAPTURE_MODE_LEFT_J 0x00
+#define ADAV80X_CAPTURE_MODE_I2S 0x01
+#define ADAV80X_CAPTURE_MODE_RIGHT_J 0x03
+#define ADAV80X_CAPTURE_MODE_MASK 0x03
+
+#define ADAV80X_PLAYBACK_MODE_MASTER 0x10
+#define ADAV80X_PLAYBACK_MODE_LEFT_J 0x00
+#define ADAV80X_PLAYBACK_MODE_I2S 0x01
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_24 0x04
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_20 0x05
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_18 0x06
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_16 0x07
+#define ADAV80X_PLAYBACK_MODE_MASK 0x07
+
+static u8 adav80x_default_regs[] = {
+ 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x02, 0x01, 0x80, 0x26, 0x00, 0x00,
+ 0x02, 0x40, 0x20, 0x00, 0x09, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x04, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0xd1, 0x92, 0xb1, 0x37,
+ 0x48, 0xd2, 0xfb, 0xca, 0xd2, 0x15, 0xe8, 0x29, 0xb9, 0x6a, 0xda, 0x2b,
+ 0xb7, 0xc0, 0x11, 0x65, 0x5c, 0xf6, 0xff, 0x8d, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xa5, 0x00, 0x00,
+ 0x00, 0xe8, 0x46, 0xe1, 0x5b, 0xd3, 0x43, 0x77, 0x93, 0xa7, 0x44, 0xee,
+ 0x32, 0x12, 0xc0, 0x11, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x3f, 0x3f,
+ 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x1d, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x52, 0x00,
+};
+
+struct adav80x {
+ enum snd_soc_control_type control_type;
+
+ enum adav80x_clk_src clk_src;
+
+ unsigned int dai_fmt[2];
+ unsigned int rate;
+ bool deemph;
+};
+
+static const char *adav80x_mux_text[] = {
+ "ADC",
+ "Playback",
+ "Aux Playback",
+};
+
+static const unsigned int adav80x_mux_values[] = {
+ 0, 2, 3,
+};
+
+#define ADAV80X_MUX_ENUM(reg, shift) \
+ SOC_VALUE_ENUM_SINGLE(reg, shift, 7, ARRAY_SIZE(adav80x_mux_text), \
+ adav80x_mux_text, adav80x_mux_values)
+
+static const struct soc_enum adav80x_mux_enum[] = {
+ ADAV80X_MUX_ENUM(ADAV80X_DPATH_CTRL1, 0),
+ ADAV80X_MUX_ENUM(ADAV80X_DPATH_CTRL1, 3),
+ ADAV80X_MUX_ENUM(ADAV80X_DPATH_CTRL2, 3),
+};
+
+static const struct snd_kcontrol_new adav80x_mux_ctrl[] = {
+ SOC_DAPM_VALUE_ENUM("Route", adav80x_mux_enum[0]),
+ SOC_DAPM_VALUE_ENUM("Route", adav80x_mux_enum[1]),
+ SOC_DAPM_VALUE_ENUM("Route", adav80x_mux_enum[2]),
+};
+
+#define ADAV80X_MUX(name, num) \
+ SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, &adav80x_mux_ctrl[num])
+
+static const struct snd_soc_dapm_widget adav80x_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("DAC", NULL, ADAV80X_DAC_CTRL1, 7, 1),
+ SND_SOC_DAPM_ADC("ADC", NULL, ADAV80X_ADC_CTRL1, 5, 1),
+
+ SND_SOC_DAPM_PGA("Right PGA", ADAV80X_ADC_CTRL1, 0, 1, NULL, 0),
+ SND_SOC_DAPM_PGA("Left PGA", ADAV80X_ADC_CTRL1, 1, 1, NULL, 0),
+
+ SND_SOC_DAPM_AIF_OUT("AIFOUT", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("AIFIN", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_AIF_OUT("AIFAUXOUT", "Aux Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("AIFAUXIN", "Aux Playback", 0, SND_SOC_NOPM, 0, 0),
+
+ ADAV80X_MUX("Aux Capture Select", 0),
+ ADAV80X_MUX("Capture Select", 1),
+ ADAV80X_MUX("DAC Select", 2),
+
+ SND_SOC_DAPM_INPUT("VINR"),
+ SND_SOC_DAPM_INPUT("VINL"),
+ SND_SOC_DAPM_OUTPUT("VOUTR"),
+ SND_SOC_DAPM_OUTPUT("VOUTL"),
+};
+
+static const struct snd_soc_dapm_route adav80x_dapm_routes[] = {
+ { "DAC Select", "ADC", "ADC" },
+ { "DAC Select", "Playback", "AIFIN" },
+ { "DAC Select", "Aux Playback", "AIFAUXIN" },
+ { "DAC", NULL, "DAC Select" },
+
+ { "Capture Select", "ADC", "ADC" },
+ { "Capture Select", "Playback", "AIFIN" },
+ { "Capture Select", "Aux Playback", "AIFAUXIN" },
+ { "AIFOUT", NULL, "Capture Select" },
+
+ { "Aux Capture Select", "ADC", "ADC" },
+ { "Aux Capture Select", "Playback", "AIFIN" },
+ { "Aux Capture Select", "Aux Playback", "AIFAUXIN" },
+ { "AIFAUXOUT", NULL, "Aux Capture Select" },
+
+ { "VOUTR", NULL, "DAC" },
+ { "VOUTL", NULL, "DAC" },
+
+ { "Left PGA", NULL, "VINL" },
+ { "Right PGA", NULL, "VINR" },
+ { "ADC", NULL, "Left PGA" },
+ { "ADC", NULL, "Right PGA" },
+};
+
+static int adav80x_set_deemph(struct snd_soc_codec *codec)
+{
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ unsigned int val;
+
+ if (adav80x->deemph) {
+ switch (adav80x->rate) {
+ case 0:
+ val = ADAV80X_DAC_CTRL2_DEEMPH_NONE;
+ break;
+ case 32000:
+ val = ADAV80X_DAC_CTRL2_DEEMPH_32;
+ break;
+ case 44100:
+ val = ADAV80X_DAC_CTRL2_DEEMPH_44;
+ break;
+ case 48000:
+ default:
+ val = ADAV80X_DAC_CTRL2_DEEMPH_48;
+ break;
+ }
+
+ } else {
+ val = ADAV80X_DAC_CTRL2_DEEMPH_NONE;
+ }
+
+ return snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2,
+ ADAV80X_DAC_CTRL2_DEEMPH_MASK, val);
+}
+
+static int adav80x_put_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ unsigned int deemph = ucontrol->value.enumerated.item[0];
+
+ if (deemph > 1)
+ return -EINVAL;
+
+ adav80x->deemph = deemph;
+
+ return adav80x_set_deemph(codec);
+}
+
+static int adav80x_get_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+ ucontrol->value.enumerated.item[0] = adav80x->deemph;
+ return 0;
+};
+
+static const DECLARE_TLV_DB_SCALE(adav80x_inpga_tlv, 0, 50, 0);
+static const DECLARE_TLV_DB_MINMAX(adav80x_digital_tlv, -9563, 0);
+
+static const struct snd_kcontrol_new adav80x_controls[] = {
+ SOC_DOUBLE_R_TLV("Master Playback Volume", ADAV80X_DAC_L_VOL,
+ ADAV80X_DAC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv),
+ SOC_DOUBLE_R_TLV("Master Capture Volume", ADAV80X_ADC_L_VOL,
+ ADAV80X_ADC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv),
+
+ SOC_DOUBLE_R_TLV("PGA Capture Volume", ADAV80X_PGA_L_VOL,
+ ADAV80X_PGA_R_VOL, 0, 0x30, 0, adav80x_inpga_tlv),
+
+ SOC_DOUBLE("Master Playback Switch", ADAV80X_DAC_CTRL1, 0, 1, 1, 0),
+ SOC_DOUBLE("Master Capture Switch", ADAV80X_ADC_CTRL1, 2, 3, 1, 1),
+
+ SOC_SINGLE("ADC High Pass Filter Switch", ADAV80X_ADC_CTRL1, 6, 1, 0),
+
+ SOC_SINGLE_BOOL_EXT("Playback De-emphasis Switch", 0,
+ adav80x_get_deemph, adav80x_put_deemph),
+};
+
+static unsigned int adav80x_port_ctrl_regs[2][2] = {
+ { ADAV80X_REC_CTRL, ADAV80X_PLAYBACK_CTRL, },
+ { ADAV80X_AUX_OUT_CTRL, ADAV80X_AUX_IN_CTRL },
+};
+
+static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ unsigned int capture = 0x00;
+ unsigned int playback = 0x00;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ capture |= ADAV80X_CAPTURE_MODE_MASTER;
+ playback |= ADAV80X_PLAYBACK_MODE_MASTER;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ capture |= ADAV80X_CAPTURE_MODE_I2S;
+ playback |= ADAV80X_PLAYBACK_MODE_I2S;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ capture |= ADAV80X_CAPTURE_MODE_LEFT_J;
+ playback |= ADAV80X_PLAYBACK_MODE_LEFT_J;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ capture |= ADAV80X_CAPTURE_MODE_RIGHT_J;
+ playback |= ADAV80X_PLAYBACK_MODE_RIGHT_J_24;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0],
+ ADAV80X_CAPTURE_MODE_MASK | ADAV80X_CAPTURE_MODE_MASTER,
+ capture);
+ snd_soc_write(codec, adav80x_port_ctrl_regs[dai->id][1], playback);
+
+ adav80x->dai_fmt[dai->id] = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+
+ return 0;
+}
+
+static int adav80x_set_adc_clock(struct snd_soc_codec *codec,
+ unsigned int sample_rate)
+{
+ unsigned int val;
+
+ if (sample_rate <= 48000)
+ val = ADAV80X_ADC_CTRL1_MODULATOR_128FS;
+ else
+ val = ADAV80X_ADC_CTRL1_MODULATOR_64FS;
+
+ snd_soc_update_bits(codec, ADAV80X_ADC_CTRL1,
+ ADAV80X_ADC_CTRL1_MODULATOR_MASK, val);
+
+ return 0;
+}
+
+static int adav80x_set_dac_clock(struct snd_soc_codec *codec,
+ unsigned int sample_rate)
+{
+ unsigned int val;
+
+ if (sample_rate <= 48000)
+ val = ADAV80X_DAC_CTRL2_DIV1 | ADAV80X_DAC_CTRL2_INTERPOL_256FS;
+ else
+ val = ADAV80X_DAC_CTRL2_DIV2 | ADAV80X_DAC_CTRL2_INTERPOL_128FS;
+
+ snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2,
+ ADAV80X_DAC_CTRL2_DIV_MASK | ADAV80X_DAC_CTRL2_INTERPOL_MASK,
+ val);
+
+ return 0;
+}
+
+static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec,
+ struct snd_soc_dai *dai, snd_pcm_format_t format)
+{
+ unsigned int val;
+
+ switch (format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ val = ADAV80X_CAPTURE_WORD_LEN16;
+ break;
+ case SNDRV_PCM_FORMAT_S18_3LE:
+ val = ADAV80X_CAPTRUE_WORD_LEN18;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ val = ADAV80X_CAPTURE_WORD_LEN20;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ val = ADAV80X_CAPTURE_WORD_LEN24;
+ break;
+ default:
+ break;
+ }
+
+ snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0],
+ ADAV80X_CAPTURE_WORD_LEN_MASK, val);
+
+ return 0;
+}
+
+static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec,
+ struct snd_soc_dai *dai, snd_pcm_format_t format)
+{
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ unsigned int val;
+
+ if (adav80x->dai_fmt[dai->id] != SND_SOC_DAIFMT_RIGHT_J)
+ return 0;
+
+ switch (format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ val = ADAV80X_PLAYBACK_MODE_RIGHT_J_16;
+ break;
+ case SNDRV_PCM_FORMAT_S18_3LE:
+ val = ADAV80X_PLAYBACK_MODE_RIGHT_J_18;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ val = ADAV80X_PLAYBACK_MODE_RIGHT_J_20;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ val = ADAV80X_PLAYBACK_MODE_RIGHT_J_24;
+ break;
+ default:
+ break;
+ }
+
+ snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1],
+ ADAV80X_PLAYBACK_MODE_MASK, val);
+
+ return 0;
+}
+
+static int adav80x_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ unsigned int rate = params_rate(params);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ adav80x_set_playback_pcm_format(codec, dai,
+ params_format(params));
+ adav80x_set_dac_clock(codec, rate);
+ } else {
+ adav80x_set_capture_pcm_format(codec, dai,
+ params_format(params));
+ adav80x_set_adc_clock(codec, rate);
+ }
+ adav80x->rate = rate;
+ adav80x_set_deemph(codec);
+
+ return 0;
+}
+
+static int adav80x_configure_pll(struct snd_soc_codec *codec, unsigned int pll,
+ int source, unsigned int freq_out)
+{
+ unsigned int pll_ctrl2 = 0;
+ unsigned int pll_src;
+
+ if (freq_out > 12288000) {
+ pll_ctrl2 |= ADAV80X_PLL_CTRL2_DOUB(pll);
+ freq_out /= 2;
+ }
+
+ /* freq_out = sample_rate * 256 */
+ switch (freq_out) {
+ case 8192000:
+ pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_32(pll);
+ break;
+ case 11289600:
+ pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_44(pll);
+ break;
+ case 12288000:
+ pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_48(pll);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (source == ADAV80X_CLK_XIN)
+ pll_src = ADAV80X_PLL_CLK_SRC_PLL_XIN(pll);
+ else
+ pll_src = ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll);
+
+ snd_soc_update_bits(codec, ADAV80X_PLL_CLK_SRC,
+ ADAV80X_PLL_CLK_SRC_PLL_MASK(pll), pll_src);
+
+ snd_soc_update_bits(codec, ADAV80X_PLL_CTRL2,
+ ADAV80X_PLL_CTRL2_PLL_MASK(pll), pll_ctrl2);
+
+ return 0;
+}
+
+static int adav80x_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ enum adav80x_clk_src new_src;
+ unsigned int pll_ctrl1 = ADAV80X_PLL_CTRL1_PLL2PD;
+
+ switch (source) {
+ case ADAV80X_PLL_SRC_XTAL:
+ case ADAV80X_PLL_SRC_XIN:
+ new_src = ADAV80X_CLK_XIN;
+ break;
+ case ADAV80X_PLL_SRC_MCLKI:
+ new_src = ADAV80X_CLK_MCLKI;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (source != ADAV80X_PLL_SRC_XTAL)
+ pll_ctrl1 |= ADAV80X_PLL_CTRL1_XTLPD;
+
+ if (freq_out) {
+ switch (freq_in) {
+ case 27000000:
+ break;
+ case 54000000:
+ pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ adav80x_configure_pll(codec, 0, new_src, freq_out);
+
+ snd_soc_write(codec, ADAV80X_PLL_CTRL1, pll_ctrl1);
+
+ if (adav80x->clk_src == ADAV80X_CLK_PLL1)
+ return 0;
+
+ adav80x->clk_src = ADAV80X_CLK_PLL1;
+
+ /* DAC, ADC, ICLK clock source - PLL1 */
+ snd_soc_write(codec, ADAV80X_ICLK_CTRL1, 0x4a);
+ snd_soc_write(codec, ADAV80X_ICLK_CTRL2, 0x10);
+
+ } else {
+ if (adav80x->clk_src == new_src)
+ return 0;
+
+ adav80x->clk_src = new_src;
+
+ if (new_src == ADAV80X_CLK_XIN) {
+ /* DAC, ADC, ICLK clock source - XIN */
+ snd_soc_write(codec, ADAV80X_ICLK_CTRL1, 0x00);
+ snd_soc_write(codec, ADAV80X_ICLK_CTRL2, 0x00);
+ } else {
+ /* DAC, ADC, ICLK clock source - MCLKI */
+ snd_soc_write(codec, ADAV80X_ICLK_CTRL1, 0x25);
+ snd_soc_write(codec, ADAV80X_ICLK_CTRL2, 0x01);
+ }
+
+ pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLL1PD;
+ snd_soc_write(codec, ADAV80X_PLL_CTRL1, pll_ctrl1);
+ }
+
+ return 0;
+}
+
+static int adav80x_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ unsigned int mask = ADAV80X_DAC_CTRL1_PD;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, 0x00);
+ break;
+ case SND_SOC_BIAS_OFF:
+ snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, mask);
+ break;
+ }
+
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+/* Enforce the same sample rate on all audio interfaces */
+static int adav80x_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+ if (!codec->active || !adav80x->rate)
+ return 0;
+
+ return snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE, adav80x->rate, adav80x->rate);
+}
+
+static void adav80x_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+ if (!codec->active)
+ adav80x->rate = 0;
+}
+
+static const struct snd_soc_dai_ops adav80x_dai_ops = {
+ .set_fmt = adav80x_set_dai_fmt,
+ .hw_params = adav80x_hw_params,
+ .set_pll = adav80x_set_dai_pll,
+ .startup = adav80x_dai_startup,
+ .shutdown = adav80x_dai_shutdown,
+};
+
+#define ADAV80X_PLAYBACK_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | \
+ SNDRV_PCM_RATE_96000)
+
+#define ADAV80X_CAPTURE_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
+
+#define ADAV80X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_driver adav80x_dais[] = {
+ {
+ .name = "adav80x-hifi",
+ .id = 0,
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = ADAV80X_PLAYBACK_RATES,
+ .formats = ADAV80X_FORMATS,
+ },
+ .capture = {
+ .stream_name = "HiFi Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = ADAV80X_CAPTURE_RATES,
+ .formats = ADAV80X_FORMATS,
+ },
+ .ops = &adav80x_dai_ops,
+ },
+ {
+ .name = "adav80x-aux",
+ .id = 1,
+ .playback = {
+ .stream_name = "Aux Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = ADAV80X_PLAYBACK_RATES,
+ .formats = ADAV80X_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Aux Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = ADAV80X_CAPTURE_RATES,
+ .formats = ADAV80X_FORMATS,
+ },
+ .ops = &adav80x_dai_ops,
+ },
+};
+
+static int adav80x_probe(struct snd_soc_codec *codec)
+{
+ int ret;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+ ret = snd_soc_codec_set_cache_io(codec, 7, 9, adav80x->control_type);
+ if (ret) {
+ dev_err(codec->dev, "failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ /* Power down SYSCLK output, power down S/PDIF receiver */
+ snd_soc_write(codec, ADAV80X_PLL_OUTE, 0x27);
+ /* Disable DAC zero flag */
+ snd_soc_write(codec, ADAV80X_DAC_CTRL3, 0x6);
+
+ return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+}
+
+static int adav80x_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+ return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static int adav80x_resume(struct snd_soc_codec *codec)
+{
+ return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+}
+
+static int adav80x_remove(struct snd_soc_codec *codec)
+{
+ return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static struct snd_soc_codec_driver adav80x_codec_driver = {
+ .probe = adav80x_probe,
+ .remove = adav80x_remove,
+ .suspend = adav80x_suspend,
+ .resume = adav80x_resume,
+ .set_bias_level = adav80x_set_bias_level,
+
+ .reg_word_size = sizeof(u8),
+ .reg_cache_size = ARRAY_SIZE(adav80x_default_regs),
+ .reg_cache_default = adav80x_default_regs,
+
+ .controls = adav80x_controls,
+ .num_controls = ARRAY_SIZE(adav80x_controls),
+ .dapm_widgets = adav80x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(adav80x_dapm_widgets),
+ .dapm_routes = adav80x_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes),
+};
+
+static int __devinit adav80x_bus_probe(struct device *dev,
+ enum snd_soc_control_type control_type)
+{
+ struct adav80x *adav80x;
+ int ret;
+
+ adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL);
+ if (!adav80x)
+ return -ENOMEM;
+
+ dev_set_drvdata(dev, adav80x);
+ adav80x->control_type = control_type;
+
+ ret = snd_soc_register_codec(dev, &adav80x_codec_driver,
+ adav80x_dais, ARRAY_SIZE(adav80x_dais));
+ if (ret)
+ kfree(adav80x);
+
+ return ret;
+}
+
+static int __devexit adav80x_bus_remove(struct device *dev)
+{
+ snd_soc_unregister_codec(dev);
+ kfree(dev_get_drvdata(dev));
+ return 0;
+}
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit adav80x_spi_probe(struct spi_device *spi)
+{
+ return adav80x_bus_probe(&spi->dev, SND_SOC_SPI);
+}
+
+static int __devexit adav80x_spi_remove(struct spi_device *spi)
+{
+ return adav80x_bus_remove(&spi->dev);
+}
+
+static struct spi_driver adav80x_spi_driver = {
+ .driver = {
+ .name = "adav801",
+ .owner = THIS_MODULE,
+ },
+ .probe = adav80x_spi_probe,
+ .remove = __devexit_p(adav80x_spi_remove),
+};
+#endif
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static const struct i2c_device_id adav80x_id[] = {
+ { "adav803", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, adav80x_id);
+
+static int __devinit adav80x_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ return adav80x_bus_probe(&client->dev, SND_SOC_I2C);
+}
+
+static int __devexit adav80x_i2c_remove(struct i2c_client *client)
+{
+ return adav80x_bus_remove(&client->dev);
+}
+
+static struct i2c_driver adav80x_i2c_driver = {
+ .driver = {
+ .name = "adav803",
+ .owner = THIS_MODULE,
+ },
+ .probe = adav80x_i2c_probe,
+ .remove = __devexit_p(adav80x_i2c_remove),
+ .id_table = adav80x_id,
+};
+#endif
+
+static int __init adav80x_init(void)
+{
+ int ret = 0;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ ret = i2c_add_driver(&adav80x_i2c_driver);
+ if (ret)
+ return ret;
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+ ret = spi_register_driver(&adav80x_spi_driver);
+#endif
+
+ return ret;
+}
+module_init(adav80x_init);
+
+static void __exit adav80x_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_del_driver(&adav80x_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&adav80x_spi_driver);
+#endif
+}
+module_exit(adav80x_exit);
+
+MODULE_DESCRIPTION("ASoC ADAV80x driver");
+MODULE_AUTHOR("Lars-Peter Clausen <[email protected]>");
+MODULE_AUTHOR("Yi Li <[email protected]>>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adav80x.h b/sound/soc/codecs/adav80x.h
new file mode 100644
index 0000000..fab23cd
--- /dev/null
+++ b/sound/soc/codecs/adav80x.h
@@ -0,0 +1,23 @@
+/*
+ * header file for ADAV80X parts
+ *
+ * Copyright 2011 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#ifndef _ADAV80X_H
+#define _ADAV80X_H
+
+enum adav80x_pll_src {
+ ADAV80X_PLL_SRC_XIN,
+ ADAV80X_PLL_SRC_XTAL,
+ ADAV80X_PLL_SRC_MCLKI,
+};
+
+enum adav80x_pll {
+ ADAV80X_PLL1 = 0,
+ ADAV80X_PLL2 = 1,
+};
+
+#endif
--
1.7.2.5
Add a machine driver to support the EVAL-ADAV801 and EVAL-ADAV803 boards
connected to a Analog Devices BF5XX evaluation board.
Signed-off-by: Lars-Peter Clausen <[email protected]>
---
sound/soc/blackfin/Kconfig | 13 +++
sound/soc/blackfin/Makefile | 2 +
sound/soc/blackfin/bfin-eval-adav80x.c | 151 ++++++++++++++++++++++++++++++++
3 files changed, 166 insertions(+), 0 deletions(-)
create mode 100644 sound/soc/blackfin/bfin-eval-adav80x.c
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index fa6a6d2..fe9d548 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -27,6 +27,19 @@ config SND_SOC_BFIN_EVAL_ADAU1701
board connected to one of the Blackfin evaluation boards like the
BF5XX-STAMP or BF5XX-EZKIT.
+config SND_SOC_BFIN_EVAL_ADAV80X
+ tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards"
+ depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
+ select SND_BF5XX_SOC_I2S
+ select SND_SOC_ADAV80X
+ help
+ Say Y if you want to add support for the Analog Devices EVAL-ADAV801 or
+ EVAL-ADAV803 board connected to one of the Blackfin evaluation boards
+ like the BF5XX-STAMP or BF5XX-EZKIT.
+
+ Note: This driver assumes that the ADAV80X digital record and playback
+ interfaces are connected to the first SPORT port on the BF5XX board.
+
config SND_BF5XX_SOC_AD73311
tristate "SoC AD73311 Audio support for Blackfin"
depends on SND_BF5XX_I2S
diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile
index f01bff6..6018bf5 100644
--- a/sound/soc/blackfin/Makefile
+++ b/sound/soc/blackfin/Makefile
@@ -22,6 +22,7 @@ snd-ssm2602-objs := bf5xx-ssm2602.o
snd-ad73311-objs := bf5xx-ad73311.o
snd-ad193x-objs := bf5xx-ad193x.o
snd-soc-bfin-eval-adau1701-objs := bfin-eval-adau1701.o
+snd-soc-bfin-eval-adav80x-objs := bfin-eval-adav80x.o
obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o
obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o
@@ -29,3 +30,4 @@ obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o
obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o
obj-$(CONFIG_SND_BF5XX_SOC_AD193X) += snd-ad193x.o
obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1701) += snd-soc-bfin-eval-adau1701.o
+obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAV80X) += snd-soc-bfin-eval-adav80x.o
diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c
new file mode 100644
index 0000000..3193e9e
--- /dev/null
+++ b/sound/soc/blackfin/bfin-eval-adav80x.c
@@ -0,0 +1,151 @@
+/*
+ * Machine driver for EVAL-ADAV801 and EVAL-ADAV803 on Analog Devices bfin
+ * evaluation boards.
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <[email protected]>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include "../codecs/adav80x.h"
+
+static const struct snd_soc_dapm_widget bfin_eval_adav80x_dapm_widgets[] = {
+ SND_SOC_DAPM_LINE("Line Out", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route bfin_eval_adav80x_dapm_routes[] = {
+ { "Line Out", NULL, "VOUTL" },
+ { "Line Out", NULL, "VOUTR" },
+
+ { "VINL", NULL, "Line In" },
+ { "VINR", NULL, "Line In" },
+};
+
+static int bfin_eval_adav80x_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dai_set_pll(codec_dai, ADAV80X_PLL1, ADAV80X_PLL_SRC_XTAL,
+ 27000000, params_rate(params) * 256);
+
+ return ret;
+}
+
+static struct snd_soc_ops bfin_eval_adav80x_ops = {
+ .hw_params = bfin_eval_adav80x_hw_params,
+};
+
+static struct snd_soc_dai_link bfin_eval_adav80x_dais[] = {
+ {
+ .name = "adav80x",
+ .stream_name = "ADAV80x HiFi",
+ .cpu_dai_name = "bfin-i2s.0",
+ .codec_dai_name = "adav80x-hifi",
+ .platform_name = "bfin-i2s-pcm-audio",
+ .ops = &bfin_eval_adav80x_ops,
+ },
+};
+
+static struct snd_soc_card bfin_eval_adav80x = {
+ .name = "bfin-eval-adav80x",
+ .dai_link = bfin_eval_adav80x_dais,
+ .num_links = ARRAY_SIZE(bfin_eval_adav80x_dais),
+
+ .dapm_widgets = bfin_eval_adav80x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adav80x_dapm_widgets),
+ .dapm_routes = bfin_eval_adav80x_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(bfin_eval_adav80x_dapm_routes),
+};
+
+enum bfin_eval_adav80x_type {
+ BFIN_EVAL_ADAV801,
+ BFIN_EVAL_ADAV803,
+};
+
+static int bfin_eval_adav80x_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &bfin_eval_adav80x;
+ const char *codec_name;
+
+ switch (platform_get_device_id(pdev)->driver_data) {
+ case BFIN_EVAL_ADAV801:
+ codec_name = "spi0.1";
+ break;
+ case BFIN_EVAL_ADAV803:
+ codec_name = "adav803.0-0034";
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ bfin_eval_adav80x_dais[0].codec_name = codec_name;
+
+ card->dev = &pdev->dev;
+
+ return snd_soc_register_card(&bfin_eval_adav80x);
+}
+
+static int __devexit bfin_eval_adav80x_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+
+ return 0;
+}
+
+static const struct platform_device_id bfin_eval_adav80x_ids[] = {
+ { "bfin-eval-adav801", BFIN_EVAL_ADAV801 },
+ { "bfin-eval-adav803", BFIN_EVAL_ADAV803 },
+ { },
+};
+MODULE_DEVICE_TABLE(platform, bfin_eval_adav80x_ids);
+
+static struct platform_driver bfin_eval_adav80x_driver = {
+ .driver = {
+ .name = "bfin-eval-adav80x",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = bfin_eval_adav80x_probe,
+ .remove = __devexit_p(bfin_eval_adav80x_remove),
+ .id_table = bfin_eval_adav80x_ids,
+};
+
+static int __init bfin_eval_adav80x_init(void)
+{
+ return platform_driver_register(&bfin_eval_adav80x_driver);
+}
+module_init(bfin_eval_adav80x_init);
+
+static void __exit bfin_eval_adav80x_exit(void)
+{
+ platform_driver_unregister(&bfin_eval_adav80x_driver);
+}
+module_exit(bfin_eval_adav80x_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <[email protected]>");
+MODULE_DESCRIPTION("ALSA SoC bfin adav80x driver");
+MODULE_LICENSE("GPL");
--
1.7.2.5
Signed-off-by: Lars-Peter Clausen <[email protected]>
---
MAINTAINERS | 1 +
1 files changed, 1 insertions(+), 0 deletions(-)
diff --git a/MAINTAINERS b/MAINTAINERS
index a26e9ec..4f7665c 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -534,6 +534,7 @@ L: [email protected] (moderated for non-subscribers)
W: http://wiki.analog.com/
S: Supported
F: sound/soc/codecs/adau*
+F: sound/soc/codecs/adav*
F: sound/soc/codecs/ad1*
F: sound/soc/codecs/ssm*
--
1.7.2.5
There is already an entry in the spi device table for the codec, but the
modalias was wrong. Also the config symbol name for the codec is wrong, so this
is fixed as well.
Signed-off-by: Lars-Peter Clausen <[email protected]>
---
arch/blackfin/mach-bf537/boards/stamp.c | 19 ++++++++++++++++---
1 files changed, 16 insertions(+), 3 deletions(-)
diff --git a/arch/blackfin/mach-bf537/boards/stamp.c b/arch/blackfin/mach-bf537/boards/stamp.c
index 5d30566..198018e 100644
--- a/arch/blackfin/mach-bf537/boards/stamp.c
+++ b/arch/blackfin/mach-bf537/boards/stamp.c
@@ -1000,9 +1000,9 @@ static struct spi_board_info bfin_spi_board_info[] __initdata = {
},
#endif
-#if defined(CONFIG_SND_BF5XX_SOC_ADAV80X) || defined(CONFIG_SND_BF5XX_SOC_ADAV80X_MODULE)
+#if defined(CONFIG_SND_SOC_ADAV80X) || defined(CONFIG_SND_SOC_ADV80X_MODULE)
{
- .modalias = "adav80x",
+ .modalias = "adav801",
.max_speed_hz = 3125000, /* max spi clock (SCK) speed in HZ */
.bus_num = 0,
.chip_select = 1,
@@ -2118,7 +2118,7 @@ static struct i2c_board_info __initdata bfin_i2c_board_info[] = {
},
#endif
-#if defined(CONFIG_SND_BF5XX_SOC_ADAV80X) || defined(CONFIG_SND_BF5XX_SOC_ADAV80X_MODULE)
+#if defined(CONFIG_SND_SOC_ADAV80X) || defined(CONFIG_SND_SOC_ADAV80X_MODULE)
{
I2C_BOARD_INFO("adav803", 0x10),
},
@@ -2553,6 +2553,14 @@ static struct platform_device bfin_ad73311_codec_device = {
};
#endif
+#if defined(CONFIG_SND_SOC_BFIN_EVAL_ADAV80X) || \
+ defined(CONFIG_SND_SOC_BFIN_EVAL_ADAV80X_MODULE)
+static struct platform_device bfin_eval_adav801_device = {
+ .name = "bfin-eval-adav801",
+ .id = -1,
+};
+#endif
+
#if defined(CONFIG_SND_BF5XX_SOC_I2S) || defined(CONFIG_SND_BF5XX_SOC_I2S_MODULE)
static struct platform_device bfin_i2s = {
.name = "bfin-i2s",
@@ -2833,6 +2841,11 @@ static struct platform_device *stamp_devices[] __initdata = {
defined(CONFIG_SND_SOC_BFIN_EVAL_ADAU1701_MODULE)
&bf5xx_adau1701_device,
#endif
+
+#if defined(CONFIG_SND_SOC_BFIN_EVAL_ADAV80X) || \
+ defined(CONFIG_SND_SOC_BFIN_EVAL_ADAV80X_MODULE)
+ &bfin_eval_adav801_device,
+#endif
};
static int __init net2272_init(void)
--
1.7.2.5
all look fine to me:
Acked-by: Mike Frysinger <[email protected]>
the Blackfin one i'll run through my tree
-mike
On Wed, Jun 22, 2011 at 11:09:54PM +0200, Lars-Peter Clausen wrote:
> diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
> new file mode 100644
> index 0000000..d2f3d08
This needs adding to MAINTAINERS.
> +static const struct soc_enum adav80x_mux_enum[] = {
> + ADAV80X_MUX_ENUM(ADAV80X_DPATH_CTRL1, 0),
> + ADAV80X_MUX_ENUM(ADAV80X_DPATH_CTRL1, 3),
> + ADAV80X_MUX_ENUM(ADAV80X_DPATH_CTRL2, 3),
> +};
No, don't use arrays...
> +static const struct snd_kcontrol_new adav80x_mux_ctrl[] = {
> + SOC_DAPM_VALUE_ENUM("Route", adav80x_mux_enum[0]),
> + SOC_DAPM_VALUE_ENUM("Route", adav80x_mux_enum[1]),
> + SOC_DAPM_VALUE_ENUM("Route", adav80x_mux_enum[2]),
> +};
...since referencing them is so error prone...
> +#define ADAV80X_MUX(name, num) \
> + SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, &adav80x_mux_ctrl[num])
...and illegible due to the magic numbers.
> + if (adav80x->deemph) {
> + switch (adav80x->rate) {
> + case 0:
> + val = ADAV80X_DAC_CTRL2_DEEMPH_NONE;
> + break;
> + case 32000:
> + val = ADAV80X_DAC_CTRL2_DEEMPH_32;
> + break;
> + case 44100:
> + val = ADAV80X_DAC_CTRL2_DEEMPH_44;
> + break;
> + case 48000:
> + default:
> + val = ADAV80X_DAC_CTRL2_DEEMPH_48;
> + break;
Really? I'd have expected a check for the closest matching rate (which
would get 32k for most low rates) or a requirement for an exact match.
> + if (freq_out) {
> + } else {
> + if (adav80x->clk_src == new_src)
> + return 0;
> +
> + adav80x->clk_src = new_src;
> +
> + if (new_src == ADAV80X_CLK_XIN) {
> + /* DAC, ADC, ICLK clock source - XIN */
> + snd_soc_write(codec, ADAV80X_ICLK_CTRL1, 0x00);
> + snd_soc_write(codec, ADAV80X_ICLK_CTRL2, 0x00);
> + } else {
> + /* DAC, ADC, ICLK clock source - MCLKI */
> + snd_soc_write(codec, ADAV80X_ICLK_CTRL1, 0x25);
> + snd_soc_write(codec, ADAV80X_ICLK_CTRL2, 0x01);
> + }
> +
> + pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLL1PD;
> + snd_soc_write(codec, ADAV80X_PLL_CTRL1, pll_ctrl1);
What's this doing? Setting the PLL output to zero means stop the PLL.
> +/* Enforce the same sample rate on all audio interfaces */
> +static int adav80x_dai_startup(struct snd_pcm_substream *substream,
> + struct snd_soc_dai *dai)
> +{
> + struct snd_soc_codec *codec = dai->codec;
> + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
> +
> + if (!codec->active || !adav80x->rate)
> + return 0;
> +
> + return snd_pcm_hw_constraint_minmax(substream->runtime,
> + SNDRV_PCM_HW_PARAM_RATE, adav80x->rate, adav80x->rate);
> +}
This means playback and capture should always run at the same rate so...
> +static struct snd_soc_dai_driver adav80x_dais[] = {
> + {
...the DAI should flag symmetric_rates, even if only for completeness.
> +static int adav80x_resume(struct snd_soc_codec *codec)
> +{
> + return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
> +}
This doesn't appear to restore the register cache, nor does
set_bias_level().
On 06/23/2011 03:21 AM, Mark Brown wrote:
> On Wed, Jun 22, 2011 at 11:09:54PM +0200, Lars-Peter Clausen wrote:
>
>> diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
>> new file mode 100644
>> index 0000000..d2f3d08
>
> This needs adding to MAINTAINERS.
Done in patch 3 of this series, but I can merge it into this one.
>> + if (adav80x->deemph) {
>> + switch (adav80x->rate) {
>> + case 0:
>> + val = ADAV80X_DAC_CTRL2_DEEMPH_NONE;
>> + break;
>> + case 32000:
>> + val = ADAV80X_DAC_CTRL2_DEEMPH_32;
>> + break;
>> + case 44100:
>> + val = ADAV80X_DAC_CTRL2_DEEMPH_44;
>> + break;
>> + case 48000:
>> + default:
>> + val = ADAV80X_DAC_CTRL2_DEEMPH_48;
>> + break;
>
> Really? I'd have expected a check for the closest matching rate (which
> would get 32k for most low rates) or a requirement for an exact match.
Since the codec supports 32k, 44.1k, 48k, 64k, 48.2k and 96k this will select
the closest match.
>
>> + if (freq_out) {
>
>> + } else {
>> + if (adav80x->clk_src == new_src)
>> + return 0;
>> +
>> + adav80x->clk_src = new_src;
>> +
>> + if (new_src == ADAV80X_CLK_XIN) {
>> + /* DAC, ADC, ICLK clock source - XIN */
>> + snd_soc_write(codec, ADAV80X_ICLK_CTRL1, 0x00);
>> + snd_soc_write(codec, ADAV80X_ICLK_CTRL2, 0x00);
>> + } else {
>> + /* DAC, ADC, ICLK clock source - MCLKI */
>> + snd_soc_write(codec, ADAV80X_ICLK_CTRL1, 0x25);
>> + snd_soc_write(codec, ADAV80X_ICLK_CTRL2, 0x01);
>> + }
>> +
>> + pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLL1PD;
>> + snd_soc_write(codec, ADAV80X_PLL_CTRL1, pll_ctrl1);
>
> What's this doing? Setting the PLL output to zero means stop the PLL.
That's exactly what's it doing. Switching to an external clock and powering the
PLL down. Or what do you mean?
>
>> +/* Enforce the same sample rate on all audio interfaces */
>> +static int adav80x_dai_startup(struct snd_pcm_substream *substream,
>> + struct snd_soc_dai *dai)
>> +{
>> + struct snd_soc_codec *codec = dai->codec;
>> + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
>> +
>> + if (!codec->active || !adav80x->rate)
>> + return 0;
>> +
>> + return snd_pcm_hw_constraint_minmax(substream->runtime,
>> + SNDRV_PCM_HW_PARAM_RATE, adav80x->rate, adav80x->rate);
>> +}
>
> This means playback and capture should always run at the same rate so...
>
>> +static struct snd_soc_dai_driver adav80x_dais[] = {
>> + {
>
> ...the DAI should flag symmetric_rates, even if only for completeness.
I had it there first, but removed it since it would mean doing the same work twice.
>
>> +static int adav80x_resume(struct snd_soc_codec *codec)
>> +{
>> + return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
>> +}
>
> This doesn't appear to restore the register cache, nor does
> set_bias_level().
The register contents is not lost unless we'd cut external power, but I could
add restoring for completeness.
On Thu, Jun 23, 2011 at 03:36:39AM +0200, Lars-Peter Clausen wrote:
> On 06/23/2011 03:21 AM, Mark Brown wrote:
> >> + if (adav80x->deemph) {
> >> + switch (adav80x->rate) {
> >> + case 0:
> >> + val = ADAV80X_DAC_CTRL2_DEEMPH_NONE;
> >> + break;
> >> + case 32000:
> >> + val = ADAV80X_DAC_CTRL2_DEEMPH_32;
> >> + break;
> >> + case 44100:
> >> + val = ADAV80X_DAC_CTRL2_DEEMPH_44;
> >> + break;
> >> + case 48000:
> >> + default:
> >> + val = ADAV80X_DAC_CTRL2_DEEMPH_48;
> >> + break;
> > Really? I'd have expected a check for the closest matching rate (which
> > would get 32k for most low rates) or a requirement for an exact match.
> Since the codec supports 32k, 44.1k, 48k, 64k, 48.2k and 96k this will select
> the closest match.
It'd be better to write the code to say that.
> >> + } else {
> >> + if (adav80x->clk_src == new_src)
> >> + return 0;
> >> +
> >> + adav80x->clk_src = new_src;
> >> +
> >> + if (new_src == ADAV80X_CLK_XIN) {
> >> + /* DAC, ADC, ICLK clock source - XIN */
> >> + snd_soc_write(codec, ADAV80X_ICLK_CTRL1, 0x00);
> >> + snd_soc_write(codec, ADAV80X_ICLK_CTRL2, 0x00);
> >> + } else {
> >> + /* DAC, ADC, ICLK clock source - MCLKI */
> >> + snd_soc_write(codec, ADAV80X_ICLK_CTRL1, 0x25);
> >> + snd_soc_write(codec, ADAV80X_ICLK_CTRL2, 0x01);
> >> + }
> >> +
> >> + pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLL1PD;
> >> + snd_soc_write(codec, ADAV80X_PLL_CTRL1, pll_ctrl1);
> > What's this doing? Setting the PLL output to zero means stop the PLL.
> That's exactly what's it doing. Switching to an external clock and powering the
> PLL down. Or what do you mean?
It really doesn't look like that - if it were just stopping the PLL it'd
not need to look at the clock source selection. Perhaps some of this
should be in set_sysclk()?
> >> +static int adav80x_resume(struct snd_soc_codec *codec)
> >> +{
> >> + return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
> >> +}
> > This doesn't appear to restore the register cache, nor does
> > set_bias_level().
> The register contents is not lost unless we'd cut external power, but I could
> add restoring for completeness.
Cutting external power is the sort of thing that one would expect to
happen over suspend on most systems, especially power sensitive ones.